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gerrit-public.fairphone.software / platform / frameworks / base / cb7e65c6cbe75c3fe55af95eba73431394669692 / media
  1. cb7e65c Better file size estimate by James Dong · 14 years ago
  2. 4c23815 Calculate audio media drift time from AudioSource by James Dong · 14 years ago
  3. a2511da Merge "Better support for buffered streaming of rtsp content, if buffer drops below a certain threshold we will temporarily pause playback until we have sufficient data." into gingerbread by Andreas Huber · 14 years ago
  4. d3c1bae Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 14 years ago
  5. 4d8f66b Better support for buffered streaming of rtsp content, if buffer drops below a certain threshold we will temporarily pause playback until we have sufficient data. by Andreas Huber · 14 years ago
  6. a87544b Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 14 years ago
  7. 6c33904 Merge "Now that AmrInputStream no longer relies on opencore, make sure it's registered in non-opencore builds." into gingerbread by Andreas Huber · 14 years ago
  8. 412fc7c Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 14 years ago
  9. 8d7d413 Now that AmrInputStream no longer relies on opencore, make sure it's registered in non-opencore builds. by Andreas Huber · 14 years ago
  10. 4dcc6a1 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 14 years ago
  11. 27b9c8e Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 14 years ago
  12. a92ebfa Audio Effects: fix problems in volume control. by Eric Laurent · 14 years ago
  13. 48ac68e Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 14 years ago
  14. e536f80 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 14 years ago
  15. 3a48d4d Better detection of connection problems - timeout if no rtcp packets arrive within a certain time, not a final frame (which may take longer) by Andreas Huber · 14 years ago
  16. 1200601 fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 14 years ago
  17. 68ae91c Merge "Recent changes to the rtsp code require every buffer fed to the packet source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder." into gingerbread by Andreas Huber · 14 years ago
  18. 0ddf8c0 Recent changes to the rtsp code require every buffer fed to the packet source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder. by Andreas Huber · 14 years ago
  19. f88ca7a0 Instead of closing the connection altogether if no UDP packets arrive after a certain time, try changing transports (to interleaved TCP). Also properly close the sockets on disconnection. by Andreas Huber · 14 years ago
  20. 681c5ff Merge "Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore." into gingerbread by Andreas Huber · 14 years ago
  21. 30cfa20 Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore. by Andreas Huber · 14 years ago
  22. 858bb4f Merge "LVM release 1.07 delivery." into gingerbread by Eric Laurent · 14 years ago
  23. f6639c4 Finetune some rtsp timeout constants. by Andreas Huber · 14 years ago
  24. df992ac Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 14 years ago
  25. c4e0b70 ALoopers can now be named (useful to distinguish threads). by Andreas Huber · 14 years ago
  26. 90862e2 Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder by James Dong · 14 years ago
  27. b86365a Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 14 years ago
  28. eeb97d9 Better support for rtsp (normal play-)time display. Better seek support, timeout if no packets arrive for too long. by Andreas Huber · 14 years ago
  29. adecf1c LVM release 1.07 delivery. by Eric Laurent · 14 years ago
  30. d6a4004 We accidentally always aborted after 10 secs, even if the connection was fine. by Andreas Huber · 14 years ago
  31. d7f1c3d Suppress the video recording start signal - bug 2950297 by James Dong · 14 years ago
  32. 17a765a Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 14 years ago
  33. 0416da7 Support for RTP packets arriving interleaved with RTSP responses. by Andreas Huber · 14 years ago
  34. 71450f8 Changed type of reverb presets from int to short by Eric Laurent · 14 years ago
  35. dfded35 Merge "Added automated tests for reverb audio effect." into gingerbread by Eric Laurent · 14 years ago
  36. 318a759 Merge "Make sure that timestamp does not go backward in MP4 file writer" into gingerbread by James Dong · 14 years ago
  37. 391e2d0 Added automated tests for reverb audio effect. by Eric Laurent · 14 years ago
  38. 8ac0983 Merge "Fix support for per-frame unsynchronization in ID3V2.4 tags." into gingerbread by Andreas Huber · 14 years ago
  39. c14f9ca Merge "Added preset reverb." into gingerbread by Eric Laurent · 14 years ago
  40. 8735f89 Fix support for per-frame unsynchronization in ID3V2.4 tags. by Andreas Huber · 14 years ago
  41. 2358402 Merge "Ensure that buffering updates eventually hit 100% after we download everything." into gingerbread by Andreas Huber · 14 years ago
  42. b8814dc Merge "Allow sniffers to return a packet of opaque data that the corresponding extractor can take advantage of to not duplicate work already done sniffing. The mp3 extractor takes advantage of this now." into gingerbread by Andreas Huber · 14 years ago
  43. efdd088 Allow sniffers to return a packet of opaque data that the corresponding extractor can take advantage of to not duplicate work already done sniffing. The mp3 extractor takes advantage of this now. by Andreas Huber · 14 years ago
  44. c23296e Ensure that buffering updates eventually hit 100% after we download everything. by Andreas Huber · 14 years ago
  45. cd295c1 Fix the simulator build. by Eric Laurent · 14 years ago
  46. a7e5648 Added preset reverb. by Eric Laurent · 14 years ago
  47. 6b6ae99 Merge "A first shot at proper support for seeking of rtsp streams." into gingerbread by Andreas Huber · 14 years ago
  48. e0dd7d3 A first shot at proper support for seeking of rtsp streams. by Andreas Huber · 14 years ago
  49. 05e80b4 Make sure that timestamp does not go backward in MP4 file writer by James Dong · 14 years ago
  50. b6d7135 Merge "LVM release 1.05 delivery" into gingerbread by Eric Laurent · 14 years ago
  51. 3e22ef1 Merge "Better handling of rtsp connection and disconnection." into gingerbread by Andreas Huber · 14 years ago
  52. 8370be1 Better handling of rtsp connection and disconnection. by Andreas Huber · 14 years ago
  53. 3f51fa7 Runtime dump support for MediaWriter by James Dong · 14 years ago
  54. b80e610 Merge "Visualizer: replace the FFT implementation with a faster one." into gingerbread by Chia-chi Yeh · 14 years ago
  55. a1a96f3 LVM release 1.05 delivery by Eric Laurent · 14 years ago
  56. fb45748 setParamMaxFileDurationUs should allow zero time input as per API of setMaxDuration. by Nipun Kwatra · 14 years ago
  57. 9767dbf Only add 4 bytes offset for the output media buffer when SPS is not received for SW AVC encoder by James Dong · 14 years ago
  58. 0ea4ed3 Don't drop a late frame which may lead to missing I frames in the MP4 file by James Dong · 14 years ago
  59. 62948fa Return error from MPEG4Writer stop() if the check on codec specific data failed by James Dong · 14 years ago
  60. cbd038f Merge "Make MediaWriter stop and pause return errors if necessary" into gingerbread by James Dong · 14 years ago
  61. d036662 Make MediaWriter stop and pause return errors if necessary by James Dong · 14 years ago
  62. a979ad6 Support for MP4V-ES packetization format according to RFC3016. by Andreas Huber · 14 years ago
  63. f0ad548 Merge "In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data." into gingerbread by Andreas Huber · 14 years ago
  64. 1aaba88 Merge "Audio Effects: fixed "strength supported" parameter size." into gingerbread by Eric Laurent · 14 years ago
  65. eef3c33 In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data. by Andreas Huber · 14 years ago
  66. 8c192fe Merge "Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description." into gingerbread by Andreas Huber · 14 years ago
  67. 58d3bd0 Visualizer: replace the FFT implementation with a faster one. by Chia-chi Yeh · 14 years ago
  68. ba8da2e Audio Effects: fixed "strength supported" parameter size. by Eric Laurent · 14 years ago
  69. af063a6 Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description. by Andreas Huber · 14 years ago
  70. 4dda6dd Merge "Make the OggExtractor less verbose." into gingerbread by Andreas Huber · 14 years ago
  71. 08c94b2 Be more lenient when validating ESDS information in mp4 audio tracks. Allow the absence of any codec specific data and assume that the mpeg4 headers are not lying to us. by Andreas Huber · 14 years ago
  72. 3386c38 Make the OggExtractor less verbose. by Andreas Huber · 14 years ago
  73. eff30e3 Change the default time scale for audio/video track during recording by James Dong · 14 years ago
  74. b720819 Use audio clock as the reference media clock by James Dong · 14 years ago
  75. e95d192 Mainly fix two mistakes that I made: by James Dong · 14 years ago
  76. 5f96138 Merge "Support getting codec, width, and height in URL for gtalk playback." into gingerbread by Mike Dodd · 14 years ago
  77. 72ac1f2 Fix software avc encoder crash at EOS. by Andreas Huber · 14 years ago
  78. 8741dfa Support getting codec, width, and height in URL for gtalk playback. by Mike Dodd · 14 years ago
  79. d790c648 Add lost preview surface detection in the JNI layer by James Dong · 14 years ago
  80. ae3a1f4 Merge "Fix the h.263 assembler to properly subset a buffer's range if it already has a range applied." into gingerbread by Andreas Huber · 14 years ago
  81. 66aa0f3 Merge "APacketSource is too verbose." into gingerbread by Andreas Huber · 14 years ago
  82. 00237b7 Fix the h.263 assembler to properly subset a buffer's range if it already has a range applied. by Andreas Huber · 14 years ago
  83. 708ec39 Don't send late frames to software encoders for encoding by James Dong · 14 years ago
  84. 45cb3cf Merge "Handle large audio lost" into gingerbread by James Dong · 14 years ago
  85. b6541f0 Merge "Fix a crash due to unnecessary check on the codec config data for H263 video track" into gingerbread by James Dong · 14 years ago
  86. 581581f Merge "Fix all fd leaks in authoring engine" into gingerbread by James Dong · 14 years ago
  87. 3f55576 APacketSource is too verbose. by Andreas Huber · 14 years ago
  88. c6280bc Fix all fd leaks in authoring engine by James Dong · 14 years ago
  89. 7ae08a6 Fix a crash due to unnecessary check on the codec config data for H263 video track by James Dong · 14 years ago
  90. 90d1d10 Merge "This code in CameraSource really should hold the lock." into gingerbread by Andreas Huber · 14 years ago
  91. 22bd242 This code in CameraSource really should hold the lock. by Andreas Huber · 14 years ago
  92. fd4a7c8 Add the a power test case which simply do the audio playback. The actual power measurement will be done by another application. by Yu Shan Emily Lau · 14 years ago
  93. afe5305 Handle large audio lost by James Dong · 14 years ago
  94. 6fa1311 Remove some obsolete code. by Andreas Huber · 14 years ago
  95. 10ed3f7 Merge "Add input buffer size check for software video encoders" into gingerbread by James Dong · 14 years ago
  96. 18f0174 Merge "We're now going to ignore timestamps completely in gtalk video conferencing, playing video as soon as it comes in. We also make up fake timestamps in the rtp code, ignoring rtcp SR information to enable early startup." into gingerbread by Andreas Huber · 14 years ago
  97. 235be39 Merge "Many, many developers misread or don't read the http specs and terminate lines with '\n' instead of CRLF '\r\n' as required. Enable the workaround for this by default. Also increase the socket read timeout to 30 secs." into gingerbread by Andreas Huber · 14 years ago
  98. f88f844 We're now going to ignore timestamps completely in gtalk video conferencing, playing video as soon as it comes in. We also make up fake timestamps in the rtp code, ignoring rtcp SR information to enable early startup. by Andreas Huber · 14 years ago
  99. e6daea5 Add input buffer size check for software video encoders by James Dong · 14 years ago
  100. f3b7859 Only check the codec specific data when the output buffer contains kKeyIsCodecConfig in MP4 writer by James Dong · 14 years ago